Voice Over Internet Protocol (VoIP) is a method for transmitting voice communication between devices through the internet. It has gained significant ground in many areas of voice transmission as it is a more efficient way of transmitting voice signals. Varying levels of compression or CODECs are used, most of which are standards such as G.729, G.723, GSM, ADPCM and others specified by the International Telephony Union (ITU).
VOIP works by taking sound and breaking that sound into digital packets. Those digital packets are transmitted to a location (based on external information transmitted as part of a call), and a receiving side translates those digital packets back into sound. Thus, each packet includes routing and sequence information. A call, or call stream, happens by doing this in two directions (duplex) between two endpoint devices.
At present, there are many VOIP equipment manufacturers worldwide including Cisco, Vocal-Tec, Clarent, Quintum, Nuera and others, which develop different types and variations of devices to transmit VOIP. Some of the manufacturers use proprietary CODECs while others use standard CODECs. Some inter-operate with other companies gateways with the most common interoperability happening with Cisco equipment, which has the largest installed base of VOIP equipment worldwide.
All of the VOIP worldwide currently is transmitted as a single call stream from one endpoint to another. There are no services which send multiple call streams of the same call through different networks or the same network in order to have an inventory of duplicate packets, thus providing an added layer of quality control by routing the call through separate environments. The inventory created by sending the same packets through multiple call streams and multiple networks allows the receiving VOIP gateway to replace a lost packet, which may occur due to internet packet loss or packets received out of order, with a duplicate of the same packet out of its inventory. This also gives the VOIP Gateway redundancy if one network is affected by a transmission interruption issue. It is unlikely for two or more separate networks not affiliated with each other at any point to go down at the same time for the same reason. Thus, by sending the same packet over multiple networks the chance of a successful call flow is statistically much higher.
There are companies which have developed different degrees of lost packet replacement methods using algorithms to predict what the lost packet would sound like (or, for non-voice packets, what information that packet would contain) based on the bordering packets (the packets before and after the lost packet). What is different with multiple call streams through multiple networks is that the endpoint may replace the lost packet with a duplicate packet from a second, third, or as many call streams through as many networks as the packets were sent. This allows the VOIP gateways to supply the end users with the full and complete call without any missing portions. The number of call streams provided may be based on the amount of actual or predicted packet loss over the network and other factors, such as whether the lost packets for a data stream occur as a percentage of the total number of packets (i.e. 1 in 10), or whether the lost packets occur in strings in a portion of the stream. The number of networks may be based on the degree in which the network operator wishes to insure quality on the call. Furthermore, with multiple network paths, the call set-up may be on one or both sides of the network.
The multiple call streams through multiple networks may work by using some of the existing CODECs or proprietary CODECs used in VOIP transmission. The software may be set to send the same call multiple times through any number of networks and tag each packet so that the same packet on each call stream is identifiable. In the case of a packet not being delivered, the system is able to identify a duplicate of that packet in another call stream. The physical layer of aspects of the invention may route multiple packets through various diverse paths or the same path. The amount of bandwidth and processing power used may be increased on a per-call basis, as compared to single-stream systems, as the endpoints may send the call multiple times at the same time and may need to handle multiple network interfaces. With this method of sending multiple packets, if all networks are running with perfect quality, then a network monitor providing routing intelligence may turn off the multiple packets without affecting the call transmission quality as the gateways are able to accept any number of call streams including one and still complete the call.